1. Field of the Invention
The present invention relates to output clipping apparatus. More specifically, the present invention relates to output clipping apparatus for hearing aids.
2. Description of the Prior Art
FIG. 1 (prior art) shows an analog hearing aid having microphone 11 connected to sound processing 12, incorporating clipping 14 and connected to power amplifier 16 and a speaker 18. Analog hearing aids clip occasionally, as it is impossible to get sufficient maximum signal level in a low power device like a hearing aid without clipping. The amplifier itself may perform the clipping function. In an analog device, the distortion caused by output clipping is acceptable, because the distortion is mostly odd order harmonics and some inter-modulation products. In a digital hearing aid output, such as that shown in FIG. 2, the effects of clipping are much worse. In a typical digital hearing aid, the circuit of FIG. 2 replaces blocks 14, 16, and 18 of FIG. 1. The clipping in such a system will create distortion products which are not harmonics or inter-modulation components, but are instead entirely unrelated to the signal, and are thus acoustically very undesirable.
One possible solution to the problem of clipping in a digital hearing aid is to convert the signal to analog, and then amplify and clip in the analog domain. This would remove the offending distortions, but at the cost of requiring greater precision in the D/A converter. As there is gain after the converter, noise will be amplified, so the noise floor would have to be better. This approach would also eliminate the possibility of a class D output stage directly in the D/A converter.
A typical digital hearing aid such as that shown in FIG. 8 includes an output digital to analog converter as one component. FIG. 2 (prior art) shows an oversampling digital to analog (D/A) converter, which utilizes a second order delta sigma quantizer 70 and a one-bit D/A converter 71 as the demodulator 69, and a low pass filter 73 to remove the noise from the one-bit signal. In one specific example of the oversampling D/A converter of FIG. 2, the input signal xi, 60, consists of data encoded into 16 bit words at 16 kHz. In a conventional D/A converter, signal 60 is clipped by clipper 61, and then placed into a register 63 from which it is fed into a low pass filter 64 at 32 kHz, with each word repeated two times. The low pass filter would typically be of the finite impulse response type. The linear interpolator 66, which is also a type of low pass filter, inserts three new words between each pair of words from low pass filter 64, which raises the data rate to 128 kHz. These words are fed into a second register 67, which feeds each word into the demodulator 69, repeating each word eight times, resulting in a data rate of 1 MHz. The 1 MHz sample rate is a sufficiently high data rate so that the quantization noise which will be introduced into the signal is small, and the requirements of the analog smoothing filter are easily met. Output yi, 61, is an analog signal.
Techniques for increasing the sample rate, generally called interpolation, are well understood by those versed in the art. Most designs will utilize several stages of increase, with each successive stage being simpler in structure, and running at a faster rate.
This sort of structure is frequently used in audio applications. The output of demodulator 69 can sometimes be driven directly into amplifier 75 and speaker 77, because the speaker can act as a low pass filter. This configuration uses what is called class D output. Power dissipation in a class D stage has the potential for being very low, as the output transistors are always in either a fully shorted or open position, removing most resistive power consumption. The remaining power is dissipated by the switching of capacitance, which is equal to C*V.sup.2 *F. C, the capacitance being switched, is typically set by the parasitic capacitance of the output transducer and of the driver transistors. V, the voltage being switched, is set by the available supplies, and the required audio output. F, the average frequency of the output, can be varied by the designer. As F is made larger, the quality of the signal improves, but the power also increases.
An over-sampling digital to analog (D/A) converter like that of FIG. 2, which includes clipping prior to the interpolating and up sampling blocks and utilizes a second order delta sigma quantizer 70, and a low pass filter 71 to convert the data from the delta sigma quantizer 70 to analog signal yi, 61, is a very effective device. However, clipping the digital signal prior to interpolating and up sampling results in a large amount of unpleasant distortion.
FIG. 3 shows a common second order delta sigma quantizer, which might be used as delta sigma quantizer 70 in FIG. 2. Delta sigma modulation incorporates a noise-shaping technique whereby the noise of a quantizer (often one-bit) operating at a frequency much greater than the bandwidth is moved to frequencies not of interest in the output signal. A filter after the quantizer removes the out of band noise. The resulting system synthesizes a high resolution data converter, but is constructed from low resolution building blocks. A good overview of the theory of delta sigma modulation is given in Oversampling Delta-Sigma Data Converters, by Candy and Temes, IEEE Press, 1992.
In practice, delta sigma modulators are generally at least second order, because higher order modulators better reduce noise in the signal band, due to improved prediction of the in-band quantization error. Thus, the resulting signal to noise ratio is better. Second order delta sigma modulators are still relatively stable, and easy to design.
Input xi, 35, is added to feedback signal 54 by adder 38. The signal from adder 38 is fed into first accumulator 40, comprising delay 42 and adder 41. The output of accumulator 40 is added to feedback signal 54 and fed into second accumulator 44, comprising delay 47 and adder 45. The output of accumulator 44 goes into quantizer 50, modeled as error signal ei, 52, added to the input by adder 51. Quantized output 36 also feeds back as feedback signal 54. Quantizer 50 may quantize the signal into ones and zeroes (one-bit format) or into multiple levels.
A need remains in the art for clipping apparatus for use with a digital hearing aid which reduces distortion.